1. Field
This invention pertains to the field of pulse width modulation (PWM) and devices, such as audio amplifiers, that process data signals using PWM.
2. Description
A switching amplifier, or class-D amplifier, is an electronic amplifier where the active devices (especially in the output stage) are operated in on/off mode (i.e., as switches). FIG. 1 shows a block diagram of one embodiment of a class-D amplifier 100 for processing an analog input signal. Amplifier 100 includes triangular wave generator 120, comparator 140, switching controller 160, and low pass filter 180. The output of amplifier 100 is provided to a relatively fixed load (e.g., a loudspeaker 10, which typically might have an impedance of 8 ohms).
Amplifier 100 employs pulse width modulation (PWM) to convey the information of the analog input signal (e.g., an audio signal). The input signal is converted to a sequence of pulses whose average value is directly proportional to the amplitude of the signal at that time. The frequency of the pulses is typically ten or more times the highest frequency of interest in the input signal. The output signal produced by switching controller 160 consists of a train of pulses whose width is a function of the amplitude & frequency of the input signal being amplified, and hence amplifier 100 is also called a PWM amplifier. The output signal from switching controller 160 is filtered by low pass filter 180 to remove the aforementioned high frequency components of the pulses. PWM amplifier 100 feeds a varying audio signal voltage into loudspeaker 10.
The output signal contains, in addition to the required amplified input signal, unwanted spectral components (i.e. the pulse frequency and its harmonics) that must be removed by low pass filter 180. Low pass filter 180 is typically fabricated using (theoretically) lossless components like inductors and capacitors in order to maintain efficiency.
FIG. 2 is a functional block diagram of one embodiment of a PWM amplifier 200. PWM amplifier 200 includes a volume control block 210, an oversampler 220, a Delta-Sigma modulator 230, a PWM mapper 240, and a filter 250.
In contrast to amplifier 100 in FIG. 1, PWM amplifier 200 operates with a digital audio input signal. It must be noted that all real world audio signals are continuous-time analog signals. Therefore, sampling and quantization must be applied to convert the continuous-time analog signal to a discrete-time digital representation for use with PWM amplifier 200.
PWM amplifier 200 receives at its input a digital audio signal as pulse-code modulated data PCM_DATA, and receives a volume control signal VOL_CON, and outputs an amplified output signal AUD_OUT. PCM is a digital representation of an analog signal where the magnitude of the signal is sampled regularly at uniform intervals, then quantized to a series of symbols in a digital (usually binary) code.
Volume control block 210 includes a volume table 211 and a multiplier 215. Volume table 211 stores in a memory volume data VOL_DATA corresponding to each value of VOL_CON. VOL_DATA is a digital code (e.g. if VOL_CON is 4-bit data→Volume Table stores 16 values for VOL_DATA). In operation, volume table 211 receives the volume control signal VOL_CON and in response thereto generates a corresponding value for VOL_DATA which it outputs as the Volume. The value of Volume is then applied to multiplier 215 in order to adjust the level of PCM_DATA to output a volume-controlled audio signal VD.
FIG. 3 illustrates a block diagram of oversampler 300 which is one possible embodiment of oversampler 220. Oversampler 300 includes a first sampler operating at a frequency Fs, a low pass interpolation filter, and a second sampler operating at a much higher frequency (e.g., 64 Fs) that the first sampler. In signal processing, oversampling is the process of sampling a signal with a sampling frequency significantly higher than twice the bandwidth or highest frequency of the signal being sampled. Oversampling reduces quantization noise and increases resolution. Oversampler 220 oversamples the volume-controlled audio signal VD which is the output by volume control block 210 and outputs an oversampled signal DSM_IN.
FIG. 4 illustrates a block diagram of Delta Sigma Modulator 400 which is one possible embodiment of Delta Sigma Modulator 230. Delta Sigma Modulator 400 includes summer 410, loop filter 420, and quantizer 430. Loop filter 420 performs noise shaping by moving the quantization noise to higher frequencies which the ear can't hear. Quantizer 430 requantizes the signal output by loop filter 420. The output of quantizer 430 is fed back to summer 410 quantizer 430 for quantization noise reduction.
Delta Sigma Modulator 230 quantizes the oversampled signal DSM_IN to produce an output signal DSM_OUT having a fewer number of bits. With current technology (e.g., a system clock of 100˜200 MHz), one can not make a PWM pulse of high resolution (e.g. 16 bits), so it needs to be re-quantized to a smaller number of bits (e.g. 4˜5 bits) by Delta-Sigma Modulator 230.
PWM mapper 240 receives the PCM signal DSM_OUT and in response thereto produces and outputs a PWM signal. PWM mapper 140 modulates the width of the pulse in the PWM signal in proportion to the volume of the input signal DSM_OUT. PWM uses a square wave whose duty cycle is modulated resulting in the variation of the average value of the waveform. FIG. 5 illustrates an operation of PWM mapper 140 in the case where a three-bit PCM signal is converted to a one-bit PWM signal.
Low Pass Filter (LPF) 250 is a filter that passes low frequency signals (i.e., the required amplified signal) and removes unwanted spectral components (i.e., signals at the pulse frequency). Beneficially, LPF 250 is made with theoretically lossless components like inductors and capacitors.
A properly designed class-D amplifier offers the following benefits: small size and weight; low power (heat) a dissipation and hence a small heatsink requirements (or no heatsink at all); low cost due to the small heat sink requirements and compact circuitry; and very high power conversion efficiency, usually ≧90%.
Hereinafter, the current which is consumed by transferring the amplified signal to the speaker is called “dynamic current” and the current which is consumed by low pass filter filtering the unwanted spectral components is called “static current.” The total current that the PWM amplifier consumes is the sum of the dynamic current and the static current.
FIG. 6 illustrates the relationship between the static current and the total current consumption in the conventional PWM amplifier. As can be seen in FIG. 6, when the amplitude of the signal (i.e., the volume of an audio signal) is at its maximum value, then the load current (i.e. the dynamic current) which is passed by the low pass filter and transferred to the load (i.e., the loudspeaker) is the greatest portion of the total current consumption of the amplifier. But as the amplitude of the signal decreases, then the total current consumption decreases while the static current consumed in the low pass filter increases so as eventually to be in excess of the load current and therefore become the greatest portion of the total current consumption of the PWM amplifier.
In practice, an audio signal is rarely set at its maximum value, and is more typically at a much lower amplitude. As a result, most of the current consumption of the PWM amplifier is attributed to the static current consumed by the low pass filter. This static current is effectively wasted power and therefore diminishes the power efficiency of the PWM amplifier.
The relationship illustrated in FIG. 6 can be explained as follows.
First, the duty ratio of the PWM signal is defined as the ratio between the period of time when the PWM signal is at the logic HIGH state and the period of time when the PWM signal is at the logic LOW state. The amount of static current in the PWM amplifier depends on the duty ratio of the PWM signal. As the duty ratio approaches 1:1, the static current increases, and as the duty ratio increases in magnitude (e.g., 1:2, 1:3 . . . ), then the static current decreases.
FIG. 7 is a flowchart illustrating operation of the conventional PWM amplifier 200. As can be seen in FIG. 7, the conventional PWM amplifier 200 maintains the duty ratio of the PWM signal close to 1:1 regardless of the volume or magnitude of the audio signal, because the audio signal is alternating between (+) and (−) values. However, as shown in FIG. 7, there are some differences in the operation of conventional PWM amplifier 200 between when the volume of the audio signal is at a maximum value and when it is not at its maximum value. When the volume of audio signal is at a maximum value, then the PWM region is fully used by the audio signal and the amount of static current is negligible as compared with dynamic current. In contrast, when the volume of the audio signal is not at a maximum value, then a portion of the PWM region is unused by the audio signal, and the amount of static current is substantial as compared with dynamic current.
FIG. 8 illustrates signals in PWM amplifier 200 in the case where the volume of the audio signal is at a maximum value. In this case, it is seen that the total range of the PWM pulse width is used by the signal.
FIG. 9 illustrates signals in PWM amplifier 200 in the case where the volume of the audio signal is not at a maximum value. In this case, it is seen that the total range of the PWM pulse width is not used by the signal.
Although the relationship between static current and dynamic current in a PWM modulator has been explained in the context of an amplifier, and particularly an audio amplifier, in general the same relationship may apply in other devices employing a PWM modulator to modulate a signal, for example, a motor control system.
Accordingly, it would be advantageous to provide a method of PWM data processing which has a reduced static current. It would also be advantageous to provide an device or system that employs pulse width modulation which exhibits a reduced static current. Other and further objects and advantages will appear hereinafter.
In one aspect of the invention, a method of processing a signal comprises: adjusting an amplitude of an input signal according to an amplitude control signal; adding an offset to the amplitude-adjusted signal to produce an offset-adjusted signal, wherein the offset is selected according to the amplitude adjustment applied to the input signal; pulse-width modulating the offset-adjusted signal to produce a pulse-width modulated signal; and filtering the pulse-width modulated signal to reduce high frequency components thereof.
In another aspect of the invention, a method of processing an input signal comprises pulse-width modulating the input signal with a pulse-width modulator (PWM) to produce a PWM signal, and then filtering the PWM signal to reduce high frequency components of the pulse-width modulated signal, further comprising adjusting a duty ratio of the PWM signal in response to an amplitude control signal.
In yet another aspect of the invention, an audio processing system comprises: a volume control adapted to adjust a volume of an input signal in response to a volume control signal; an offset generator adapted to generate an offset to be applied to the volume-adjusted input signal, wherein the offset is selected in response to the volume control signal; a combiner adapted to apply the offset to the volume-adjusted input signal to produce an offset-adjusted signal; a pulse width modulator adapted to pulse-width modulate the offset-adjusted signal; and a filter adapted to reduce high frequency components of the pulse-width modulated signal.
In still another aspect of the invention, a motor control system comprises: an amplitude control adapted to adjust an amplitude of an input signal in response to an amplitude control signal; an offset generator adapted to generate an offset to be applied to the amplitude-adjusted input signal, wherein the offset is selected in response to the amplitude control signal; a combiner adapted to apply the offset to the amplitude-adjusted input signal to produce an offset-adjusted signal; a pulse width modulator adapted to pulse-width modulate the offset-adjusted signal; and a filter adapted to reduce high frequency components of the pulse-width modulated signal.
In a further aspect of the invention, a system adapted to process an input signal with a pulse-width modulator (PWM) to produce a PWM signal, and further adapted to filter the PWM signal to reduce high frequency components of the PWM signal, further comprises a duty-cycle adjustment element adapted to adjust a duty cycle of the PWM signal in response to an amplitude control signal.